[Freeswitch-users] Configuration

[freeswitch-users at lists.freeswitch.org]

Hello

I still got problems with the configuration, since the new layout was added.

Today I've updated to latest trunk (config files too) and I've configured
the system.

I've only done the following changes:
1. I've set domain=mydomain.com in vars.xml
2. I've renamed sip_profiles/default to mydomain.com and nat to
nat_mydomain.com
3. I've enabled mod_xml_curl and configured it to fetch dial plans
4. I've commented out mod_dialplan_asterisk and mod_iax
5. I've added my extensions to directory\default (copied brian's config and
made appropriate changes)

My sip gateway do not use authentication (and I therefore just use "
number at mygateway.com" in the dialplan for external calls).
The phones register by using mydomain.com as proxy/registrar (not by using
an ip).

The UA's can register fine and external outbound calls works gr8.

Current problems:

* I only get audio one way when dialing from one registered UA to another,
check attached log internal_call.log

* Inbound external calls do not work, i get [407][Proxy Authentication
Required], check log external_call.log

//Jonas
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